A recent development in the field of telephony has been the introduction of digital subscriber line (DSL) techniques whereby data traffic is transported over a subscriber loop to a customer. The subscriber line is coupled to a DSL line card in the local exchange whereby the DSL traffic is coupled to the subscriber line. Typically, this traffic is transported at least in the downstream direction to the customer in a discrete multi-tone (DMT) format in which the information is spread over a number of carriers.
There is increasing interest in Internet access and a continual demand to increase speed of delivery of Internet traffic to end users. For residential users it is impractical to install a large scale fibre network to allow high speed access to individual homes. Therefore there is a strong financial incentive to reuse the existing copper pair connections between the home and a central office or local exchange rather than to incur the expense of installation of new customer connections. Various systems have been devised to support higher data transmission speeds over these pairs. In particular, ADSL (asymmetric digital subscriber line) and other similar systems are under development and are currently being defined in the standards forum. A feature of ADSL is that the transmission rates in the upstream and downstream directions are asymmetric. The data rate in the downstream direction from the line card towards the customer is much higher than the upstream rate, which is typically be only 150–200 kbit/s.
There is also growing interest in transmitting conventional POTS voice traffic using packet oriented data networks. A natural progression is to use the ADSL connection to provide one or more voice connections to the customer, either in addition to or instead of the conventional baseband POTS connection. Additional POTS channels delivered in this way are known as ‘derived POTS’ channels. Preferably, the voice traffic is multiplexed with data traffic using the same ADSL link, this being referred to as voice over DSL. Attempts to introduce such a service have not however been entirely successful as it has been that the high quality of service requirements for voice traffic are difficult to achieve with current techniques.
Carrier class voice requires certain quality criteria to be met. In particular, the end to end delay experienced by the voice signal must be kept to a minimum to avoid echo and, in extreme cases, delays large enough to disrupt two way conversation.
Conventionally, IP traffic streams are multiplexed by interleaving complete packets. Since IP data packets can be quite long (an Ethernet packet could for example be up to 1500 bytes in length) a high priority voice packet might well have to wait a significant time before a transmission opportunity arises. This is a particular problem when the data transmission rate is low, such as in the upstream direction on an ADSL link. This low transmission rate is a consequence of the serial nature and the limited bandwidth of the upstream path. At a typical upstream rate of 200 kbit/s, a 1500 byte packet takes 60 msec to transmit. Thus, a voice packet might be delayed by up to 60 msec. This is a major source of delay in a packet voice network and is unacceptable for high quality voice traffic.
There is a proposal to the IETF to use a multiplexing scheme whereby long IP packets can be interrupted by short, high priority packets such as voice. This interruption is achieved by fooling the HDLC layer, which is a transmission layer below IP, that the IP data packet has finished, and then resuming the original data packet transmission after the high priority packet has been sent. However, this system is specific to links using HDLC and requires modifications to the HDLC protocol layer at each end of the link. It is thus not readily adaptable to general ADSL use.
A second potential source of delay arises from the time taken to collect enough digitised voice samples to fill a voice packet. Conventionally, when using standard telephony pulse code modulation (64 kbit/s), a voice sample is produced every 125 microseconds, each sample being represented by one byte. To maintain transmission efficiency, it is desirable that the payload of a packet should be relatively long compared to its header which conveys essential addressing information but otherwise incorporates no useful revenue earning data. In a system based on Internet protocol (IP), voice traffic would normally be sent using UDP and RTP protocol layers above IP. Each layer requires its own header which contains information related to the corresponding layer. The total length of such a header can be at least 40 bytes. Thus, for 75% efficiency (say) the payload length should be 120 bytes. Using 64 kbit/s PCM this implies a packetisation time of 15 msec. This is again undesirable for efficient voice transport.